Remove compatibility things

Possible with bumped FFmpeg/Libav.

These are just the simple cases.

5 files changed, 1 insertions, 75 deletions

diff --git a/audio/decode/ad_lavc.c b/audio/decode/ad_lavc.c
index e28558414d..276ce69739 100644
--- a/audio/decode/ad_lavc.c
+++ b/audio/decode/ad_lavc.c
@@ -207,7 +207,6 @@ static int decode_packet(struct dec_audio *da, struct demux_packet *mpkt,
     if (priv->needs_reset)
         control(da, ADCTRL_RESET, NULL);
 
-#if HAVE_AVCODEC_NEW_CODEC_API
     int ret = avcodec_send_packet(avctx, &pkt);
     if (ret >= 0 || ret == AVERROR(EAGAIN) || ret == AVERROR_EOF) {
         if (ret >= 0 && mpkt)
@@ -220,24 +219,6 @@ static int decode_packet(struct dec_audio *da, struct demux_packet *mpkt,
         if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
             ret = 0;
     }
-#else
-    int ret = avcodec_decode_audio4(avctx, priv->avframe, &got_frame, &pkt);
-    if (mpkt) {
-        // At least "shorten" decodes sub-frames, instead of the whole packet.
-        // At least "mpc8" can return 0 and wants the packet again next time.
-        if (ret >= 0) {
-            ret = FFMIN(ret, mpkt->len); // sanity check against decoder overreads
-            mpkt->buffer += ret;
-            mpkt->len    -= ret;
-            mpkt->pts = MP_NOPTS_VALUE; // don't reset PTS next time
-        }
-        // LATM may need many packets to find mux info
-        if (ret == AVERROR(EAGAIN)) {
-            mpkt->len = 0;
-            return 0;
-        }
-    }
-#endif
     if (ret < 0) {
         MP_ERR(da, "Error decoding audio.\n");
         return -1;
@@ -245,8 +226,7 @@ static int decode_packet(struct dec_audio *da, struct demux_packet *mpkt,
     if (!got_frame)
         return 0;
 
-    double out_pts = mp_pts_from_av(MP_AVFRAME_DEC_PTS(priv->avframe),
-                                    &priv->codec_timebase);
+    double out_pts = mp_pts_from_av(priv->avframe->pts, &priv->codec_timebase);
 
     struct mp_audio *mpframe = mp_audio_from_avframe(priv->avframe);
     if (!mpframe)
diff --git a/audio/decode/ad_spdif.c b/audio/decode/ad_spdif.c
index 56e4a8102d..e15aca5c53 100644
--- a/audio/decode/ad_spdif.c
+++ b/audio/decode/ad_spdif.c
@@ -116,16 +116,10 @@ static int determine_codec_profile(struct dec_audio *da, AVPacket *pkt)
         goto done;
     }
 
-#if HAVE_AVCODEC_NEW_CODEC_API
     if (avcodec_send_packet(ctx, pkt) < 0)
         goto done;
     if (avcodec_receive_frame(ctx, frame) < 0)
         goto done;
-#else
-    int got_frame = 0;
-    if (avcodec_decode_audio4(ctx, frame, &got_frame, pkt) < 1 || !got_frame)
-        goto done;
-#endif
 
     profile = ctx->profile;
 
@@ -178,11 +172,7 @@ static int init_filter(struct dec_audio *da, AVPacket *pkt)
     if (!stream)
         goto fail;
 
-#if HAVE_AVCODEC_HAS_CODECPAR
     stream->codecpar->codec_id = spdif_ctx->codec_id;
-#else
-    stream->codec->codec_id = spdif_ctx->codec_id;
-#endif
 
     AVDictionary *format_opts = NULL;
 
diff --git a/audio/filter/af_lavcac3enc.c b/audio/filter/af_lavcac3enc.c
index 0a7c5d4440..9df5adb96f 100644
--- a/audio/filter/af_lavcac3enc.c
+++ b/audio/filter/af_lavcac3enc.c
@@ -280,7 +280,6 @@ static int filter_out(struct af_instance *af)
     AVPacket pkt = {0};
     av_init_packet(&pkt);
 
-#if HAVE_AVCODEC_NEW_CODEC_API
     // Send input as long as it wants.
     while (1) {
         err = read_input_frame(af, frame);
@@ -310,21 +309,6 @@ static int filter_out(struct af_instance *af)
         MP_FATAL(af, "Encode failed.\n");
         goto done;
     }
-#else
-    err = read_input_frame(af, frame);
-    if (err < 0)
-        goto done;
-    if (err == 0)
-        goto done;
-    err = -1;
-    int ok;
-    int lavc_ret = avcodec_encode_audio2(s->lavc_actx, &pkt, frame, &ok);
-    s->input->samples = 0;
-    if (lavc_ret < 0 || !ok) {
-        MP_FATAL(af, "Encode failed.\n");
-        goto done;
-    }
-#endif
 
     MP_DBG(af, "avcodec_encode_audio got %d, pending %d.\n",
            pkt.size, s->pending->samples + s->input->samples);
diff --git a/audio/filter/af_lavrresample.c b/audio/filter/af_lavrresample.c
index dc5d1a0d23..828be66247 100644
--- a/audio/filter/af_lavrresample.c
+++ b/audio/filter/af_lavrresample.c
@@ -111,12 +111,7 @@ static double get_delay(struct af_resample *s)
 }
 static int get_out_samples(struct af_resample *s, int in_samples)
 {
-#if LIBSWRESAMPLE_VERSION_MAJOR > 1 || LIBSWRESAMPLE_VERSION_MINOR >= 2
     return swr_get_out_samples(s->avrctx, in_samples);
-#else
-    return av_rescale_rnd(in_samples, s->out_rate, s->in_rate, AV_ROUND_UP)
-           + swr_get_delay(s->avrctx, s->out_rate);
-#endif
 }
 #endif
 
diff --git a/audio/out/ao_lavc.c b/audio/out/ao_lavc.c
index 8ae1317407..4dbc55a369 100644
--- a/audio/out/ao_lavc.c
+++ b/audio/out/ao_lavc.c
@@ -258,7 +258,6 @@ static void encode_audio_and_write(struct ao *ao, AVFrame *frame)
     struct priv *ac = ao->priv;
     AVPacket packet = {0};
 
-#if HAVE_AVCODEC_NEW_CODEC_API
     int status = avcodec_send_frame(ac->codec, frame);
     if (status < 0) {
         MP_ERR(ao, "error encoding at %d %d/%d\n",
@@ -297,28 +296,6 @@ static void encode_audio_and_write(struct ao *ao, AVFrame *frame)
         write_packet(ao, &packet);
         av_packet_unref(&packet);
     }
-#else
-    av_init_packet(&packet);
-    int got_packet = 0;
-    int status = avcodec_encode_audio2(ac->codec, &packet, frame, &got_packet);
-    if (status < 0) {
-        MP_ERR(ao, "error encoding at %d %d/%d\n",
-               frame ? (int) frame->pts : -1,
-               ac->codec->time_base.num,
-               ac->codec->time_base.den);
-        return;
-    }
-    if (!got_packet) {
-        return;
-    }
-    if (frame) {
-        if (ac->savepts == AV_NOPTS_VALUE)
-            ac->savepts = frame->pts;
-    }
-    encode_lavc_write_stats(ao->encode_lavc_ctx, ac->codec);
-    write_packet(ao, &packet);
-    av_packet_unref(&packet);
-#endif
 }
 
 // must get exactly ac->aframesize amount of data